The threshold of audibility of phase noise in ADC and DAC clocks is a fairly contentious issue in the HiFi and audiophile world. Some sources claim that jitter is clearly audible at low levels, and some claim that high levels of jitter are inaudible. The literature describes several tests, many with conflicting results.

One of the chief difficulties in testing the audibility of jitter is that it requires a complex hardware setup, which means that many listeners would be required to be present for an time consuming (and expensive) on site test. Over the last couple of months I have been thinking about organising a distributed listening test to look at the audibility of jitter in audio applications, based on algorithms for simulating the effects of jitter on signals. These algorithms are fairly well described in RF and telecomms engineering literature, and would be interesting for comparison purposes.

The kind of thing I have in mind is this:Use samples which are accepted to sound good -> simulate jitter -> perform listening tests -> perform more tests at different levels of jitter depending on results

The purpose of this thread is to get ideas of the Hydrogenaudio community about performing these tests. Some of the things I would appreciate input on are:

File 2 and 3 null out almost perfectly. Amplification of that difference by 50 and then another 20 dB* produces a beautiful example of a quiet muffled song, drowning in a thick ocean of noise. Can we reject any chance of audible difference here right off the bat?

Also, I'm not sure why there are 4 files on offer, but only 2 inverted files. I didn't use the inverted ones. Was I supposed to compare 2/3 and 4/5?

4 and 5 have a tiny subsample difference, so the diff is a complete piece of music, but without the bass. I don't have the qualifications to explain why the bass region vanishes.

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like if you didn't really care that much.

The warbled tone extends rather far into the sample, and I suppose it was a basis for Arnold to dismiss the samples immediately. I couldn't ABX any of it, in any case.

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Your 05 file is a fraction of a sample earlier than your 04 file

How did you know 5 is earlier than 4? Better tools?

*) Audacity apparently doesn't allow more than 50dB amplification in a single step.

4 and 5 have a tiny subsample difference, so the diff is a complete piece of music, but without the bass. I don't have the qualifications to explain why the bass region vanishes.

Taking the delta between samples a fixed time interval apart acts as a low-pass filter with a 6 dB roll off, until the period approaches the time delta. It then acts as a notch filter when the period and the time delta are equal.

4 and 5 have a tiny subsample difference, so the diff is a complete piece of music, but without the bass. I don't have the qualifications to explain why the bass region vanishes.

Taking the delta between samples a fixed time interval apart acts as a low-pass filter with a 6 dB roll off, until the period approaches the time delta. It then acts as a notch filter when the period and the time delta are equal.

Is this right? You're describing the comb filter you get when you add suitably delayed signals together. However, when you subtract delayed signals, I think the basic response is a high pass filter - and if the delay is increased to give a comb filter effect, the lowest frequencies are always attenuated - i.e. DC is always in a notch filter.

(don't take my word for this - far too sleep deprived to be 100% sure of anything today )

4 and 5 have a tiny subsample difference, so the diff is a complete piece of music, but without the bass. I don't have the qualifications to explain why the bass region vanishes.

Taking the delta between samples a fixed time interval apart acts as a low-pass filter with a 6 dB roll off, until the period approaches the time delta. It then acts as a notch filter when the period and the time delta are equal.

Is this right? You're describing the comb filter you get when you add suitably delayed signals together. However, when you subtract delayed signals, I think the basic response is a high pass filter - and if the delay is increased to give a comb filter effect, the lowest frequencies are always attenuated - i.e. DC is always in a notch filter.

(don't take my word for this - far too sleep deprived to be 100% sure of anything today )

Cheers,David.

Oops...I meant high pass.

BTW, if the period is twice the delay then the signal is amplified 6 dB.